Merge changes from team/group/appdocsxml

This commit introduces the first phase of an effort to manage documentation of the
interfaces in Asterisk in an XML format.  Currently, a new format is available for
applications and dialplan functions.  A good number of conversions to the new format
are also included.

For more information, see the following message to asterisk-dev:

http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Russell Bryant
2008-11-01 21:10:07 +00:00
parent 1fef0f63bb
commit 5b168ee34b
111 changed files with 8063 additions and 2478 deletions

View File

@@ -62,188 +62,396 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/global_datastores.h"
#include "asterisk/dsp.h"
/*** DOCUMENTATION
<application name="Dial" language="en_US">
<synopsis>
Attempt to connect to another device or endpoint and bridge the call.
</synopsis>
<syntax>
<parameter name="Technology/Resource" required="true" argsep="&amp;">
<argument name="Technology/Resource" required="true">
<para>Specification of the device(s) to dial. These must be in the format of
<literal>Technology/Resource</literal>, where <replaceable>Technology</replaceable>
represents a particular channel driver, and <replaceable>Resource</replaceable>
represents a resource available to that particular channel driver.</para>
</argument>
<argument name="Technology2/Resource2" required="false" multiple="true">
<para>Optional extra devices to dial in parallel</para>
<para>If you need more then one enter them as
Technology2/Resource2&amp;Technology3/Resourse3&amp;.....</para>
</argument>
</parameter>
<parameter name="timeout" required="false">
<para>Specifies the number of seconds we attempt to dial the specified devices</para>
<para>If not specified, this defaults to 136 years.</para>
</parameter>
<parameter name="options" required="false">
<optionlist>
<option name="A">
<argument name="x" required="true">
<para>The file to play to the called party</para>
</argument>
<para>Play an announcement to the called party, where <replaceable>x</replaceable> is the prompt to be played</para>
</option>
<option name="C">
<para>Reset the call detail record (CDR) for this call.</para>
</option>
<option name="c">
<para>If the Dial() application cancels this call, always set the flag to tell the channel
driver that the call is answered elsewhere.</para>
</option>
<option name="d">
<para>Allow the calling user to dial a 1 digit extension while waiting for
a call to be answered. Exit to that extension if it exists in the
current context, or the context defined in the <variable>EXITCONTEXT</variable> variable,
if it exists.</para>
</option>
<option name="D" argsep=":">
<argument name="called" />
<argument name="calling" />
<para>Send the specified DTMF strings <emphasis>after</emphasis> the called
party has answered, but before the call gets bridged. The
<replaceable>called</replaceable> DTMF string is sent to the called party, and the
<replaceable>calling</replaceable> DTMF string is sent to the calling party. Both arguments
can be used alone.</para>
</option>
<option name="e">
<para>Execute the <literal>h</literal> extension for peer after the call ends</para>
</option>
<option name="f">
<para>Force the callerid of the <emphasis>calling</emphasis> channel to be set as the
extension associated with the channel using a dialplan <literal>hint</literal>.
For example, some PSTNs do not allow CallerID to be set to anything
other than the number assigned to the caller.</para>
</option>
<option name="F" argsep="^">
<argument name="context" required="false" />
<argument name="exten" required="false" />
<argument name="priority" required="true" />
<para>When the caller hangs up, transfer the called party
to the specified destination and continue execution at that location.</para>
</option>
<option name="g">
<para>Proceed with dialplan execution at the next priority in the current extension if the
destination channel hangs up.</para>
</option>
<option name="G" argsep="^">
<argument name="context" required="false" />
<argument name="exten" required="false" />
<argument name="priority" required="true" />
<para>If the call is answered, transfer the calling party to
the specified <replaceable>priority</replaceable> and the called party to the specified
<replaceable>priority</replaceable> plus one.</para>
<note>
<para>You cannot use any additional action post answer options in conjunction with this option.</para>
</note>
</option>
<option name="h">
<para>Allow the called party to hang up by sending the <literal>*</literal> DTMF digit.</para>
</option>
<option name="H">
<para>Allow the calling party to hang up by hitting the <literal>*</literal> DTMF digit.</para>
</option>
<option name="i">
<para>Asterisk will ignore any forwarding requests it may receive on this dial attempt.</para>
</option>
<option name="k">
<para>Allow the called party to enable parking of the call by sending
the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
</option>
<option name="K">
<para>Allow the calling party to enable parking of the call by sending
the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
</option>
<option name="L" argsep=":">
<argument name="x" required="true">
<para>Maximum call time, in milliseconds</para>
</argument>
<argument name="y">
<para>Warning time, in milliseconds</para>
</argument>
<argument name="z">
<para>Repeat time, in milliseconds</para>
</argument>
<para>Limit the call to <replaceable>x</replaceable> milliseconds. Play a warning when <replaceable>y</replaceable> milliseconds are
left. Repeat the warning every <replaceable>z</replaceable> milliseconds until time expires.</para>
<para>This option is affected by the following variables:</para>
<variablelist>
<variable name="LIMIT_PLAYAUDIO_CALLER">
<value name="yes" default="true" />
<value name="no" />
<para>If set, this variable causes Asterisk to play the prompts to the caller.</para>
</variable>
<variable name="LIMIT_PLAYAUDIO_CALLEE">
<value name="yes" />
<value name="no" default="true"/>
<para>If set, this variable causes Asterisk to play the prompts to the callee.</para>
</variable>
<variable name="LIMIT_TIMEOUT_FILE">
<value name="filename"/>
<para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the timeout is reached.
If not set, the time remaining will be announced.</para>
</variable>
<variable name="LIMIT_CONNECT_FILE">
<value name="filename"/>
<para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the call begins.
If not set, the time remaining will be announced.</para>
</variable>
<variable name="LIMIT_WARNING_FILE">
<value name="filename"/>
<para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play as
a warning when time <replaceable>x</replaceable> is reached. If not set, the time remaining will be announced.</para>
</variable>
</variablelist>
</option>
<option name="m">
<argument name="class" required="false"/>
<para>Provide hold music to the calling party until a requested
channel answers. A specific music on hold <replaceable>class</replaceable>
(as defined in <filename>musiconhold.conf</filename>) can be specified.</para>
</option>
<option name="M" argsep="^">
<argument name="macro" required="true">
<para>Name of the macro that should be executed.</para>
</argument>
<argument name="arg" multiple="true">
<para>Macro arguments</para>
</argument>
<para>Execute the specified <replaceable>macro</replaceable> for the <emphasis>called</emphasis> channel
before connecting to the calling channel. Arguments can be specified to the Macro
using <literal>^</literal> as a delimiter. The macro can set the variable
<variable>MACRO_RESULT</variable> to specify the following actions after the macro is
finished executing:</para>
<variablelist>
<variable name="MACRO_RESULT">
<para>If set, this action will be taken after the macro finished executing.</para>
<value name="ABORT">
Hangup both legs of the call
</value>
<value name="CONGESTION">
Behave as if line congestion was encountered
</value>
<value name="BUSY">
Behave as if a busy signal was encountered
</value>
<value name="CONTINUE">
Hangup the called party and allow the calling party to continue dialplan execution at the next priority
</value>
<!-- TODO: Fix this syntax up, once we've figured out how to specify the GOTO syntax -->
<value name="GOTO:&lt;context&gt;^&lt;exten&gt;^&lt;priority&gt;">
Transfer the call to the specified destination.
</value>
</variable>
</variablelist>
<note>
<para>You cannot use any additional action post answer options in conjunction
with this option. Also, pbx services are not run on the peer (called) channel,
so you will not be able to set timeouts via the TIMEOUT() function in this macro.</para>
</note>
</option>
<option name="n">
<para>This option is a modifier for the call screening/privacy mode. (See the
<literal>p</literal> and <literal>P</literal> options.) It specifies
that no introductions are to be saved in the <directory>priv-callerintros</directory>
directory.</para>
</option>
<option name="N">
<para>This option is a modifier for the call screening/privacy mode. It specifies
that if Caller*ID is present, do not screen the call.</para>
</option>
<option name="o">
<para>Specify that the Caller*ID that was present on the <emphasis>calling</emphasis> channel
be set as the Caller*ID on the <emphasis>called</emphasis> channel. This was the
behavior of Asterisk 1.0 and earlier.</para>
</option>
<option name="O">
<argument name="mode">
<para>With <replaceable>mode</replaceable> either not specified or set to <literal>1</literal>,
the originator hanging up will cause the phone to ring back immediately.</para>
<para>With <replaceable>mode</replaceable> set to <literal>2</literal>, when the operator
flashes the trunk, it will ring their phone back.</para>
</argument>
<para>Enables <emphasis>operator services</emphasis> mode. This option only
works when bridging a DAHDI channel to another DAHDI channel
only. if specified on non-DAHDI interfaces, it will be ignored.
When the destination answers (presumably an operator services
station), the originator no longer has control of their line.
They may hang up, but the switch will not release their line
until the destination party (the operator) hangs up.</para>
</option>
<option name="p">
<para>This option enables screening mode. This is basically Privacy mode
without memory.</para>
</option>
<option name="P">
<argument name="x" />
<para>Enable privacy mode. Use <replaceable>x</replaceable> as the family/key in the AstDB database if
it is provided. The current extension is used if a database family/key is not specified.</para>
</option>
<option name="r">
<para>Indicate ringing to the calling party, even if the called party isn't actually ringing. Pass no audio to the calling
party until the called channel has answered.</para>
</option>
<option name="S">
<argument name="x" required="true" />
<para>Hang up the call <replaceable>x</replaceable> seconds <emphasis>after</emphasis> the called party has
answered the call.</para>
</option>
<option name="t">
<para>Allow the called party to transfer the calling party by sending the
DTMF sequence defined in <filename>features.conf</filename>.</para>
</option>
<option name="T">
<para>Allow the calling party to transfer the called party by sending the
DTMF sequence defined in <filename>features.conf</filename>.</para>
</option>
<option name="U" argsep="^">
<argument name="x" required="true">
<para>Name of the subroutine to execute via Gosub</para>
</argument>
<argument name="arg" multiple="true" required="false">
<para>Arguments for the Gosub routine</para>
</argument>
<para>Execute via Gosub the routine <replaceable>x</replaceable> for the <emphasis>called</emphasis> channel before connecting
to the calling channel. Arguments can be specified to the Gosub
using <literal>^</literal> as a delimiter. The Gosub routine can set the variable
<variable>GOSUB_RESULT</variable> to specify the following actions after the Gosub returns.</para>
<variablelist>
<variable name="GOSUB_RESULT">
<value name="ABORT">
Hangup both legs of the call.
</value>
<value name="CONGESTION">
Behave as if line congestion was encountered.
</value>
<value name="BUSY">
Behave as if a busy signal was encountered.
</value>
<value name="CONTINUE">
Hangup the called party and allow the calling party
to continue dialplan execution at the next priority.
</value>
<!-- TODO: Fix this syntax up, once we've figured out how to specify the GOTO syntax -->
<value name="GOTO:&lt;context&gt;^&lt;exten&gt;^&lt;priority&gt;">
Transfer the call to the specified priority. Optionally, an extension, or
extension and priority can be specified.
</value>
</variable>
</variablelist>
<note>
<para>You cannot use any additional action post answer options in conjunction
with this option. Also, pbx services are not run on the peer (called) channel,
so you will not be able to set timeouts via the TIMEOUT() function in this routine.</para>
</note>
</option>
<option name="w">
<para>Allow the called party to enable recording of the call by sending
the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
</option>
<option name="W">
<para>Allow the calling party to enable recording of the call by sending
the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
</option>
<option name="x">
<para>Allow the called party to enable recording of the call by sending
the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
</option>
<option name="X">
<para>Allow the calling party to enable recording of the call by sending
the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
</option>
</optionlist>
</parameter>
<parameter name="URL">
<para>The optional URL will be sent to the called party if the channel driver supports it.</para>
</parameter>
</syntax>
<description>
<para>This application will place calls to one or more specified channels. As soon
as one of the requested channels answers, the originating channel will be
answered, if it has not already been answered. These two channels will then
be active in a bridged call. All other channels that were requested will then
be hung up.</para>
<para>Unless there is a timeout specified, the Dial application will wait
indefinitely until one of the called channels answers, the user hangs up, or
if all of the called channels are busy or unavailable. Dialplan executing will
continue if no requested channels can be called, or if the timeout expires.
This application will report normal termination if the originating channel
hangs up, or if the call is bridged and either of the parties in the bridge
ends the call.</para>
<para>If the <variable>OUTBOUND_GROUP</variable> variable is set, all peer channels created by this
application will be put into that group (as in Set(GROUP()=...).
If the <variable>OUTBOUND_GROUP_ONCE</variable> variable is set, all peer channels created by this
application will be put into that group (as in Set(GROUP()=...). Unlike OUTBOUND_GROUP,
however, the variable will be unset after use.</para>
<para>This application sets the following channel variables:</para>
<variablelist>
<variable name="DIALEDTIME">
<para>This is the time from dialing a channel until when it is disconnected.</para>
</variable>
<variable name="ANSWEREDTIME">
<para>This is the amount of time for actual call.</para>
</variable>
<variable name="DIALSTATUS">
<para>This is the status of the call</para>
<value name="CHANUNAVAIL" />
<value name="CONGESTION" />
<value name="NOANSWER" />
<value name="BUSY" />
<value name="ANSWER" />
<value name="CANCEL" />
<value name="DONTCALL">
For the Privacy and Screening Modes.
Will be set if the called party chooses to send the calling party to the 'Go Away' script.
</value>
<value name="TORTURE">
For the Privacy and Screening Modes.
Will be set if the called party chooses to send the calling party to the 'torture' script.
</value>
<value name="INVALIDARGS" />
</variable>
</variablelist>
</description>
</application>
<application name="RetryDial" language="en_US">
<synopsis>
Place a call, retrying on failure allowing an optional exit extension.
</synopsis>
<syntax>
<parameter name="announce" required="true">
<para>Filename of sound that will be played when no channel can be reached</para>
</parameter>
<parameter name="sleep" required="true">
<para>Number of seconds to wait after a dialattempt failed before a new attempt is made</para>
</parameter>
<parameter name="retries" required="true">
<para>Number of retries</para>
<para>When this is reached flow will continue at the next priority in the dialplan</para>
</parameter>
<parameter name="dialargs" required="true">
<para>Same format as arguments provided to the Dial application</para>
</parameter>
</syntax>
<description>
<para>This application will attempt to place a call using the normal Dial application.
If no channel can be reached, the <replaceable>announce</replaceable> file will be played.
Then, it will wait <replaceable>sleep</replaceable> number of seconds before retrying the call.
After <replaceable>retries</replaceable> number of attempts, the calling channel will continue at the next priority in the dialplan.
If the <replaceable>retries</replaceable> setting is set to 0, this application will retry endlessly.
While waiting to retry a call, a 1 digit extension may be dialed. If that
extension exists in either the context defined in <variable>EXITCONTEXT</variable> or the current
one, The call will jump to that extension immediately.
The <replaceable>dialargs</replaceable> are specified in the same format that arguments are provided
to the Dial application.</para>
</description>
</application>
***/
static char *app = "Dial";
static char *synopsis = "Place a call and connect to the current channel";
static char *descrip =
" Dial(Technology/resource[&Tech2/resource2...][,timeout][,options][,URL]):\n"
"This application will place calls to one or more specified channels. As soon\n"
"as one of the requested channels answers, the originating channel will be\n"
"answered, if it has not already been answered. These two channels will then\n"
"be active in a bridged call. All other channels that were requested will then\n"
"be hung up.\n"
" Unless there is a timeout specified, the Dial application will wait\n"
"indefinitely until one of the called channels answers, the user hangs up, or\n"
"if all of the called channels are busy or unavailable. Dialplan executing will\n"
"continue if no requested channels can be called, or if the timeout expires.\n\n"
" This application sets the following channel variables upon completion:\n"
" DIALEDTIME - This is the time from dialing a channel until when it\n"
" is disconnected.\n"
" ANSWEREDTIME - This is the amount of time for actual call.\n"
" DIALSTATUS - This is the status of the call:\n"
" CHANUNAVAIL | CONGESTION | NOANSWER | BUSY | ANSWER | CANCEL\n"
" DONTCALL | TORTURE | INVALIDARGS\n"
" For the Privacy and Screening Modes, the DIALSTATUS variable will be set to\n"
"DONTCALL if the called party chooses to send the calling party to the 'Go Away'\n"
"script. The DIALSTATUS variable will be set to TORTURE if the called party\n"
"wants to send the caller to the 'torture' script.\n"
" This application will report normal termination if the originating channel\n"
"hangs up, or if the call is bridged and either of the parties in the bridge\n"
"ends the call.\n"
" The optional URL will be sent to the called party if the channel supports it.\n"
" If the OUTBOUND_GROUP variable is set, all peer channels created by this\n"
"application will be put into that group (as in Set(GROUP()=...).\n"
" If the OUTBOUND_GROUP_ONCE variable is set, all peer channels created by this\n"
"application will be put into that group (as in Set(GROUP()=...). Unlike OUTBOUND_GROUP,\n"
"however, the variable will be unset after use.\n\n"
" Options:\n"
" A(x) - Play an announcement to the called party, using 'x' as the file.\n"
" C - Reset the CDR for this call.\n"
" c - If DIAL cancels this call, always set the flag to tell the channel\n"
" driver that the call is answered elsewhere.\n"
" d - Allow the calling user to dial a 1 digit extension while waiting for\n"
" a call to be answered. Exit to that extension if it exists in the\n"
" current context, or the context defined in the EXITCONTEXT variable,\n"
" if it exists.\n"
" D([called][:calling]) - Send the specified DTMF strings *after* the called\n"
" party has answered, but before the call gets bridged. The 'called'\n"
" DTMF string is sent to the called party, and the 'calling' DTMF\n"
" string is sent to the calling party. Both parameters can be used\n"
" alone.\n"
" e - execute the 'h' extension for peer after the call ends. This\n"
" operation will not be performed if the peer was parked\n"
" f - Force the callerid of the *calling* channel to be set as the\n"
" extension associated with the channel using a dialplan 'hint'.\n"
" For example, some PSTNs do not allow CallerID to be set to anything\n"
" other than the number assigned to the caller.\n"
" F(context^exten^pri) - When the caller hangs up, transfer the called party\n"
" to the specified context and extension and continue execution.\n"
" g - Proceed with dialplan execution at the current extension if the\n"
" destination channel hangs up.\n"
" G(context^exten^pri) - If the call is answered, transfer the calling party to\n"
" the specified priority and the called party to the specified priority+1.\n"
" Optionally, an extension, or extension and context may be specified. \n"
" Otherwise, the current extension is used. You cannot use any additional\n"
" action post answer options in conjunction with this option.\n"
" h - Allow the called party to hang up by sending the '*' DTMF digit, or\n"
" whatever sequence was defined in the featuremap section for\n"
" 'disconnect' in features.conf\n"
" H - Allow the calling party to hang up by hitting the '*' DTMF digit, or\n"
" whatever sequence was defined in the featuremap section for\n"
" 'disconnect' in features.conf\n"
" i - Asterisk will ignore any forwarding requests it may receive on this\n"
" dial attempt.\n"
" k - Allow the called party to enable parking of the call by sending\n"
" the DTMF sequence defined for call parking in the featuremap section of features.conf.\n"
" K - Allow the calling party to enable parking of the call by sending\n"
" the DTMF sequence defined for call parking in the featuremap section of features.conf.\n"
" L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when 'y' ms are\n"
" left. Repeat the warning every 'z' ms. The following special\n"
" variables can be used with this option:\n"
" * LIMIT_PLAYAUDIO_CALLER yes|no (default yes)\n"
" Play sounds to the caller.\n"
" * LIMIT_PLAYAUDIO_CALLEE yes|no\n"
" Play sounds to the callee.\n"
" * LIMIT_TIMEOUT_FILE File to play when time is up.\n"
" * LIMIT_CONNECT_FILE File to play when call begins.\n"
" * LIMIT_WARNING_FILE File to play as warning if 'y' is defined.\n"
" The default is to say the time remaining.\n"
" m([class]) - Provide hold music to the calling party until a requested\n"
" channel answers. A specific MusicOnHold class can be\n"
" specified.\n"
" M(x[^arg]) - Execute the Macro for the *called* channel before connecting\n"
" to the calling channel. Arguments can be specified to the Macro\n"
" using '^' as a delimiter. The Macro can set the variable\n"
" MACRO_RESULT to specify the following actions after the Macro is\n"
" finished executing.\n"
" * ABORT Hangup both legs of the call.\n"
" * CONGESTION Behave as if line congestion was encountered.\n"
" * BUSY Behave as if a busy signal was encountered.\n"
" * CONTINUE Hangup the called party and allow the calling party\n"
" to continue dialplan execution at the next priority.\n"
" * GOTO:<context>^<exten>^<priority> - Transfer the call to the\n"
" specified priority. Optionally, an extension, or\n"
" extension and priority can be specified.\n"
" You cannot use any additional action post answer options in conjunction\n"
" with this option. Also, pbx services are not run on the peer (called) channel,\n"
" so you will not be able to set timeouts via the TIMEOUT() function in this macro.\n"
" n - This option is a modifier for the screen/privacy mode. It specifies\n"
" that no introductions are to be saved in the priv-callerintros\n"
" directory.\n"
" N - This option is a modifier for the screen/privacy mode. It specifies\n"
" that if callerID is present, do not screen the call.\n"
" o - Specify that the CallerID that was present on the *calling* channel\n"
" be set as the CallerID on the *called* channel. This was the\n"
" behavior of Asterisk 1.0 and earlier.\n"
" O([x]) - \"Operator Services\" mode (DAHDI channel to DAHDI channel\n"
" only, if specified on non-DAHDI interface, it will be ignored).\n"
" When the destination answers (presumably an operator services\n"
" station), the originator no longer has control of their line.\n"
" They may hang up, but the switch will not release their line\n"
" until the destination party hangs up (the operator). Specified\n"
" without an arg, or with 1 as an arg, the originator hanging up\n"
" will cause the phone to ring back immediately. With a 2 specified,\n"
" when the \"operator\" flashes the trunk, it will ring their phone\n"
" back.\n"
" p - This option enables screening mode. This is basically Privacy mode\n"
" without memory.\n"
" P([x]) - Enable privacy mode. Use 'x' as the family/key in the database if\n"
" it is provided. The current extension is used if a database\n"
" family/key is not specified.\n"
" r - Indicate ringing to the calling party. Pass no audio to the calling\n"
" party until the called channel has answered.\n"
" S(x) - Hang up the call after 'x' seconds *after* the called party has\n"
" answered the call.\n"
" t - Allow the called party to transfer the calling party by sending the\n"
" DTMF sequence defined in the blindxfer setting in the featuremap section\n"
" of features.conf.\n"
" T - Allow the calling party to transfer the called party by sending the\n"
" DTMF sequence defined in the blindxfer setting in the featuremap section\n"
" of features.conf.\n"
" U(x[^arg]) - Execute via Gosub the routine 'x' for the *called* channel before connecting\n"
" to the calling channel. Arguments can be specified to the Gosub\n"
" using '^' as a delimiter. The Gosub routine can set the variable\n"
" GOSUB_RESULT to specify the following actions after the Gosub returns.\n"
" * ABORT Hangup both legs of the call.\n"
" * CONGESTION Behave as if line congestion was encountered.\n"
" * BUSY Behave as if a busy signal was encountered.\n"
" * CONTINUE Hangup the called party and allow the calling party\n"
" to continue dialplan execution at the next priority.\n"
" * GOTO:<context>^<exten>^<priority> - Transfer the call to the\n"
" specified priority. Optionally, an extension, or\n"
" extension and priority can be specified.\n"
" You cannot use any additional action post answer options in conjunction\n"
" with this option. Also, pbx services are not run on the peer (called) channel,\n"
" so you will not be able to set timeouts via the TIMEOUT() function in this routine.\n"
" w - Allow the called party to enable recording of the call by sending\n"
" the DTMF sequence defined in the automon setting in the featuremap section\n"
" of features.conf.\n"
" W - Allow the calling party to enable recording of the call by sending\n"
" the DTMF sequence defined in the automon setting in the featuremap section\n"
" of features.conf.\n"
" x - Allow the called party to enable recording of the call by sending\n"
" the DTMF sequence defined in the automixmon setting in the featuremap section\n"
" of features.conf.\n"
" X - Allow the calling party to enable recording of the call by sending\n"
" the DTMF sequence defined in the automixmon setting in the featuremap section\n"
" of features.conf.\n";
/* RetryDial App by Anthony Minessale II <anthmct@yahoo.com> Jan/2005 */
static char *rapp = "RetryDial";
static char *rsynopsis = "Place a call, retrying on failure allowing optional exit extension.";
static char *rdescrip =
" RetryDial(announce,sleep,retries,dialargs): This application will attempt to\n"
"place a call using the normal Dial application. If no channel can be reached,\n"
"the 'announce' file will be played. Then, it will wait 'sleep' number of\n"
"seconds before retrying the call. After 'retries' number of attempts, the\n"
"calling channel will continue at the next priority in the dialplan. If the\n"
"'retries' setting is set to 0, this application will retry endlessly.\n"
" While waiting to retry a call, a 1 digit extension may be dialed. If that\n"
"extension exists in either the context defined in ${EXITCONTEXT} or the current\n"
"one, The call will jump to that extension immediately.\n"
" The 'dialargs' are specified in the same format that arguments are provided\n"
"to the Dial application.\n";
enum {
OPT_ANNOUNCE = (1 << 0),
@@ -2187,8 +2395,8 @@ static int load_module(void)
else
ast_add_extension2(con, 1, "s", 1, NULL, NULL, "KeepAlive", ast_strdup(""), ast_free_ptr, "app_dial");
res = ast_register_application(app, dial_exec, synopsis, descrip);
res |= ast_register_application(rapp, retrydial_exec, rsynopsis, rdescrip);
res = ast_register_application_xml(app, dial_exec);
res |= ast_register_application_xml(rapp, retrydial_exec);
return res;
}